[asterisk-users] Polycom 601 and Paging

[asterisk at txpe.net]

My sip.cfg (v.1.6.5) has the following lines:

<alertInfo voIpProt.SIP.alertInfo.1.value="" 
voIpProt.SIP.alertInfo.1.class=""/>
There is not a "2" version.

and
<RING_ANSWER se.rt.4.name="Ring Answer" se.rt.4.type="ring-answer" 
se.rt.4.timeout="2000" se.rt.4.ringer="2" se.rt.4.callWait="6" 
se.rt.4.mod="1"/>

At 04:27 PM 7/18/2006, you wrote:
>I cant do step 2.
>I cant find:
>
>2. Okay, see how the SIPAddHeader includes "Ring Answer"? That word 
>or words will be matched by alertInfo in sip.cfg in order to figure 
>out what to do. You are using the config files from krisk.org listed 
>above, right? If not, go get them now. I'll wait. So in sip.cfg in 
>the <voIpProt><SIP> section you need a line like:
>
><alertInfo voIpProt.SIP.alertInfo.2.value="Ring Answer" 
>voIpProt.SIP.alertInfo.2.class="4"/>
>
>----- Original Message ----- From: "Brian Vincent (C)" 
><VincentB at coppercolorado.com>
>To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
><asterisk-users at lists.digium.com>
>Sent: Tuesday, July 18, 2006 5:09 PM
>Subject: RE: [asterisk-users] Polycom 601 and Paging
>
>
>
>I have these instructions on the wiki in the comments section.  I 
>had a hard time following the directions too, but I finally got it to work:
>
>We've got 3 things going on with setting up Auto Answer and Ring 
>Answer. Let's detail this process from beginning to end using Ring 
>Answer as our example. (Auto Answer isn't much different except you 
>want to make sure step #2 below goes to class 3 rather than 4, and 
>that class 3 is set up as described elsewhere which is the same as 
>the one in the ipmid.cfg file from krisk.org.)
>
>1. First, use the SIPAddHeader() directive in Asterisk to properly 
>alert the phone. In my situation, I have 10 phones with 2-digit 
>extensions. I want to call each phone by prefixing the extension 
>with a "1" in order to activate the intercom. For example, if I dial 
>126 I want it to put extension 26 on speakerphone. So go into 
>extensions.conf and make sure you create a new section like this:
><a href='icm-auto-answer'>icm-auto-answer </a href='icm-auto-answer'>
>;intercom
>exten => _12x,1,SIPAddHeader(Alert-Info: Ring Answer)
>exten => _12x,2,Dial(sip/${EXTEN:1:3})
>exten => _12x,3,Hangup
>exten => _12x,102,Hangup
>
>Then make sure in your from-internal section of extensions.conf you 
>have a include => icm-auto-answer
>
>2. Okay, see how the SIPAddHeader includes "Ring Answer"? That word 
>or words will be matched by alertInfo in sip.cfg in order to figure 
>out what to do. You are using the config files from krisk.org listed 
>above, right? If not, go get them now. I'll wait. So in sip.cfg in 
>the <voIpProt><SIP> section you need a line like:
>
><alertInfo voIpProt.SIP.alertInfo.2.value="Ring Answer" 
>voIpProt.SIP.alertInfo.2.class="4"/>
>
>The value parameter must match whatever you use in the SIPAddHeader 
>string. In this case they're both "Ring Answer". You could just as 
>easily replace both with the word "Foo" or "RA".
>
>3. Now, the alertInfo tag will match that value and then go to the 
>"class" value to figure out what to do. Se we need to make sure 
>class="4" is set up properly. You could probably set up class 4 in 
>sip.cfg, but mine lives in ipmid.cfg. So go into ipmid.cfg and 
>locate the <ringtypes> section. Below that tag (and before it's 
>corresponding </ringtype> closing tag) you need to make sure class 4 
>is set up right. You should have this line:
><RING_ANSWER se.rt.4.name="Ring Answer" se.rt.4.type="ring-answer" 
>se.rt.4.timeout="1000" se.rt.4.ringer="2" se.rt.4.callWait="6" 
>se.rt.4.mod="1"/>
>
>The notes above describe that line. The key is that this is class 4 
>as noted by the 3rd part of the value names - se.rt.4.name. I'd like 
>to add that the keyword "RING_ANSWER" is meaningless, it's just a 
>human-readable tag.
>
>Got all that? The SIPAddHeader of "Ring Answer" hits the <alertInfo> 
>tag to figure out which class to go to. Then the class in ipmid.cfg 
>says, "Oh, I'm a "ring-answer" type and my firmware knows what to do 
>with that type."
>
>One test you can do is to connect to asterisk ($ asterisk -r), bump 
>your verbosity up (<tt>set verbose 6</tt>), and try to place a call 
>using that context from step #1. You'll see one phone calling 
>another and within the Asterisk CLI you should see the following 
>message appear:
>- Executing SIPAddHeader("SIP/20-86bc", "Alert-Info: Ring Answer") 
>in new stack<br />
>Extension Changed 20 new state InUse for Notify User 26<br />
>- Executing Dial("SIP/20-86bc", "sip/26") in new stack<br />
>- Called 26<br />
>- SIP/26-0448 is ringing<br />
>- SIP/26-0448 answered SIP/20-86bc<br />
>- Attempting native bridge of SIP/20-86bc and SIP/26-0448
>
>If you don't see that Alert-Info: Ring Answer being sent, then you 
>know you haven't gotten the first step right.
>
>Also, I made the mistake of putting some comments into the .cfg 
>files and the comments seemed to screw up the parser. It ignored 
>seemingly random lines (i.e. non-comment ones). I'm not a complete 
>moron since I've been writing XML for 6 years (and HTML for 11) but 
>it goes to show how careful you should be. Anyway, I use "xmllint" 
>on config files now before rebooting the phones to make sure I 
>didn't make a dumb typo.
>-------------------
>Brian Vincent
>Copper Mountain Telecom
>vincentb at coppercolorado.com